1. Field of the Invention
The present invention generally relates to a method and a mobile phone terminal enabling to merge telephony services over heterogeneous networks, i. e. a circuit switched network and a packet switched data network. It concerns all the mobile terminals that can be used for telephony, that is to say: phone handsets, smart phones, and personal computers comprising a telephony interface.
2. Description of the Prior Art
In an enterprise, a unified communication server provides sophisticated voice services to employees of this enterprise, such as unified directories research, communication logs, instant messaging, telephonic and instant messaging presence, and voice services such as call hold, back and forth, conference, call forward, access to central directory, dial by name, call log, voice mail, . . . . These services are fully available for fixed telephone terminals, but are only partially available for mobile phone terminals when these mobile terminals are outside the premises of the enterprise.
In particular, a unified communication server offers some basic services to the classical GSM handsets, such as conference, call forwarding, business voice mail consulting, dual call, business forwarding activation and deactivation, etc. These basic services are activated through DTMF codes sent over the GSM circuit switched network. DTMF codes respectively represent the actions of a user on the twelve dial keys of a phone. These codes can transmit twelve symbols only, and they cannot be used for backwards transmission of data, from the server to the terminals, because the mobile phone terminals generally do not comprise DTMF decoders.
It is desirable to extend these services, by adding more sophisticated services such as: business directory consulting, business call log consulting, dial by name, instant messaging, telephony presence, etc, that a unified communication classically offers to fixed terminals. These sophisticated services cannot be activated through Dual Tone Multi Frequency (DTMF) codes sent over the GSM circuit switched, because they need more signaling data and a bidirectional transmission. They are implemented by a signaling data transmission via a mobile packet switched data network, such as a 3G network (GPRS, UMTS, HSDPA, . . . ) or high data rate wireless local area network (WiFi, WiMAX, . . . ), or a newer packet switched network.
Current voice application solutions for mobile handsets are deployed either in a public land mobile circuit switched network (with basic services only) or in a public land packet switched data network (with sophisticated services in addition to the basic services), but never in both type of networks at the same time. That is to say that signaling is either carried over a circuit switched network (GSM for instance) or over a packet switched data network (GPRS, UMTS, HSDPA, . . . ) but is never carried on both networks.
It is desirable to alternately use both types of networks to offer, to the end users, sophisticated telephony services whenever it is possible, and to offer only basic services, as a backup solution, when it is not possible to offer the sophisticated services (No network coverage).
A service could generate signaling data on a circuit switched network and on a packet data network, at the same time. However, in such a case, both signaling types should be handled, though they are sent asynchronously. For instance, when setting up a call, the voice application should wait for the circuit switched call signaling, to consider that a voice link is really active. On the other hand, packet data network signaling may be lost, depending on the network coverage. So it would be necessary to synchronize signaling data coming from both types of networks when a service is activated, and to provide a backup mode when only one or several circuit switched networks are available.
Thus, there is a need to provide a technical solution to synchronize signaling data coming from both types of networks when a service is activated. The 3GPP community has developed a solution supported by the Internet Multimedia Subsystem (IMS). However this solution has two drawbacks: The IMS is not widely deployed today, and it will not be deployed in the enterprises because it would be too expensive.
The document WO 2006/137762 describes a terminal in a mobile communication system arranged for transmitting a first and a second media stream (respectively for video and voice) to a receiving terminal, the first and the second media stream being transmitted separate from each other, but at least partly simultaneously. The first media stream is associated with a first end-to-end time delay and the second media stream is associated with a second end-to-end time delay. The first end-to-end time delay is larger than the second end-to-end time delay.
The terminal further comprises:                control means arranged for presenting the first media stream at the terminal delayed from the transmission of the first media stream with a time dependent on the time difference between the first and the second end-to-end time delay;        and means for estimating end-to-end time delays arranged for estimating the first end-to-end time delay and the second end-to-end time delay.        
The first end-to-end time delay is estimated by:                recording a first time when a data packet to which a reply is requested is transmitted to the receiving terminal;        recording a second time when the reply is received from the receiving terminal;        subtracting the first time from the second time;        optionally, subtracting, from the result of the subtraction, a possible waiting time that the receiving terminal has waited from receiving the transmitted data packet until the reply was sent to the terminal;        dividing the result of the subtracting calculation by two, and;        adding, to the result of the division calculation, a current buffer delay in the receiving terminal, which buffer delay is measured at the receiving terminal and transmitted to the sending terminal.        
This known solution is not adapted for synchronizing first and second data indicating events concerning a same call, because:
These events do not generate a continuous flow of data, as voice and video do, because they are scarce. So it is not possible to estimate a time shift once for all. Restituting accurately the time intervals between the events is not important for events concerning a call. Most important is correctly restituting the order of the perception of events, because it is essential for establishing or releasing a call, and all other call services.
The aim of the present invention is to solve this synchronization problem by simple means.